sorcery: Create sorcery instance registry.

In order to retrieve an arbitrary sorcery instance from a dialplan function
(or any place else) there needs to be a registry of sorcery instances.

ast_sorcery_init now creates a hashtab as a registry.

ast_sorcery_open now checks the hashtab for an existing sorcery instance
matching the caller's module name.  If it finds one, it bumps the 
refcount and returns it.  If not, it creates a new sorcery instance,
adds it to the hashtab, then returns it.

ast_sorcery_retrieve_by_module_name is a new function that does a hashtab 
lookup by module name.  It can be called by the future dialplan function.

res_pjsip/config_system needed a small change to share the main res_pjsip 
sorcery instance.

tests/test_sorcery was updated to include a test for the registry.

(closes issue ASTERISK-22537)
Review: http://reviewboard.asterisk.org/r/3184/
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git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@408519 f38db490-d61c-443f-a65b-d21fe96a405b
pjsip_cli: Fix memory leak in ast_sip_cli_print_sorcery_objectset.

Fixed memory leaks in ast_sip_cli_print_sorcery_objectset and
ast_variable_list_sort.  

(closes issue ASTERISK-23266)
Review: http://reviewboard.asterisk.org/r/3200/
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git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@408521 f38db490-d61c-443f-a65b-d21fe96a405b
pjsip_cli: Add pjsip commands 'show registrations' and 'show contacts'.

Added 'show registrations' and 'show contacts' to pjsip cli to make things
a little more consistent.  The output is exactly the same as the list command.

Just needed to add entries to their respective ast_cli_entry structures.

(closes issue ASTERISK-23275)
Review: http://reviewboard.asterisk.org/r/3210/
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git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@408523 f38db490-d61c-443f-a65b-d21fe96a405b
app_chanspy: Documentation Update To Clarify "x" Option

When using the "x" option (specify a DTMF digit to exit the application), it is
not obvious in the documentation that this only works when spying on a channel.
If a channel being used to spy on other channels is waiting to connect to a
channel or is no longer attached to a channel, the DTMF is ignored.

As noted on the issue tracker, since there are workarounds available and this is
a rarely used option we are opting for a documentation change here.

(closes issue ASTERISK-22661)
Reported by: Chris Hillman
Patches:
    asterisk-22661-doc-clarify-chan_spy.diff uploaded by Michael L. Young

Review: https://reviewboard.asterisk.org/r/2990/
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git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@408539 f38db490-d61c-443f-a65b-d21fe96a405b
Fix type of roundTripDelay variables
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git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@408592 f38db490-d61c-443f-a65b-d21fe96a405b
channel.c: MOH is not working for transferee after attended transfer

Updated the code to check to see if MOH is playing on the transferor and if
so then start it on the channel that replaces it during a masquerade.

Example scenario of the problem:
Alice calls Bob and then Bob begins the attended transfer process into a queue.
Upon going on hold Alice hears music and so does Bob once he is in the queue.
Bob then transfers Alice into the queue and then music for Alice stops even
though she should be hearing it since has now replaced Bob in the queue.

The problem that was occurring is that once the channel was masqueraded the app
(queues, confbridge, etc...) had no way of knowing that the channel had just
been swapped out thus it did not start music for the present channel.

Credit to Olle Johansson for pointing me in the right direction on this issue.

(closes issue ASTERISK-19499)
Reported by: Timo Teräs
Review: https://reviewboard.asterisk.org/r/3226/
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git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@408645 f38db490-d61c-443f-a65b-d21fe96a405b
rtp_engine: Output mixup in ${CHANNEL(rtpqos,audio,all)}

Fixed the output of CHANNEL(rtpqos,audio,all) to use txjitter instead
of rxjitter.

(closes issue ASTERISK-23261)
Reported by: rsw686
Patches:
     rtpqos.patch uploaded by rsw686 (license 5887)
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git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@408652 f38db490-d61c-443f-a65b-d21fe96a405b
chan_sip: prevent add_route from adding empty header.

Fix regression caused by ASTERISK-22582.  Empty Route
headers were added when the route had a single strict
hop.

(closes issue ASTERISK-23306)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3236/


git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@408699 f38db490-d61c-443f-a65b-d21fe96a405b
json: Fix json API wrapper code for json library versions earlier than 2.3.0.

* Fixed json ref counting issue with json API wrapper code for
ast_json_object_update_existing() and ast_json_object_update_missing()
when the json library is earlier than version 2.3.0.
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git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@408712 f38db490-d61c-443f-a65b-d21fe96a405b
json: Fix off-nominal json ref counting issues.

* Fixed off-nominal json ref counting issue with using the following API
calls: ast_json_object_set() and ast_json_array_append().

* Fixed off-nominal error reporting in ast_ari_endpoints_list().

* Fixed some miscellaneous off-nominal json ref counting issues in
report_receive_fax_status() and dial_to_json().
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git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@408714 f38db490-d61c-443f-a65b-d21fe96a405b
manager: Fix AMI Status action of a single channel.

Fixed use of uninitialized ao2 container iterator in an off-nominal
condition.  Either a memory allocation error or the requested channel is
an internal channel not exposed to the outside.
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git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@408723 f38db490-d61c-443f-a65b-d21fe96a405b
rtp_engine: Dynamic payload change in rtp mapping not supported

Asterisk didn't support the dynamic payload change in rtp mapping in the 200
OK response.

Scenario:
Asterisk sends the INVITE proposing alaw and telephone-event, it proposes
rtpmap:101 for telephone-event.  Peer responds with 2xx, it answers with
alaw and telephone-event also, but it proposes a different rtpmap number
(rtpmap:103) for telephone-event.

Expected Behaviour:
Asterisk should honour the rtpmapping in the response and send DTMF packets
using 103 as payload type for DTMF.

Actual Behaviour: Asterisk sends DTMF packets using payload type 101.

With this patch asterisk now supports changes that can occur in the rtp mapping
in the response.

(closes issue ASTERISK-23279)
Reported by: NITESH BANSAL
Review: https://reviewboard.asterisk.org/r/3225/
Patches:
     dynamic_payload_change.patch uploaded by nbansal (license 6418)
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git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@408731 f38db490-d61c-443f-a65b-d21fe96a405b
Remove extra defines of AST_PBX_MAX_STACK.

* Ensure AST_PBX_MAX_STACK is only defined in extconf.h and pbx.h.
* Fix incorrect function parameters in utils/extconf.c.

(closes issue ASTERISK-23141)
Reported by: Maxim
Review: https://reviewboard.asterisk.org/r/3241/
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git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@408788 f38db490-d61c-443f-a65b-d21fe96a405b
ignore AST_CONTROL_PVT_CAUSE_CODE without any messages

(closes issue ASTERISK-23336)
Reported by: Alexander Semych
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git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@408840 f38db490-d61c-443f-a65b-d21fe96a405b
main: Initialize dialplan providing core components prior to module pre-load

It is possible to pre-load pbx_config. As a result, pbx_config - which will
load and parse the dialplan - will attempt to use various dialplan components,
such as device state providers and presence state providers, prior to them
being initialized by the core. This would lead to a crash, as the components
had not created their Stasis cache entries.

This patch moves a number of core component initializations before the module
pre-load. This guarantees that if someone does pre-load pbx_config - or other
pbx modules - that the Stasis caches for the various core components are
created.

(closes issue ASTERISK-23320)
Reported by: xrobau

(closes issue ASTERISK-23265)
Reported by: Andrew Nagy
Tested by: Andrew Nagy, Rusty Newton
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